Digital Audio

La presente investigación exploró el papel de las mediaciones tecnológicas, psicoacústicas y socioculturales, involucradas en siete experiencias de creación sonora, tanto individuales como colaborativas, bajo las posturas estéticas del paisaje sonoro y del glitch. Además de un recorrido histórico por el desarrollo de éstas, se produjeron algunas ideas sobre su escucha y sus procesos creativos. El sustento teórico de la investigación se apoyó en las ideas de artistas y personas académicas cuyo campo de estudio refiere al papel y uso de los medios en la creación sonora con dichas posturas, los postulados acerca del ritornelo y el cuerpo sin órganos de Deleuze y Guattari, y las nociones sobre lo post-digital. Ahora bien, el concepto de medio se empleó desde su acepción en las ideas de McLuhan, así como también en tanto producto anclado a la normalización social y diversos intereses económicos y políticos que fomentan su invisibilización. Se analizaron a los micrófonos, las bocinas, el software de audio digital, los formatos de archivos digitales y las diversas formas de licencia y distribución en internet, así como a algunas variables biológicas, psicológicas y sociales propias de la especie humana, que intervienen en la creación sonora mediada por el trabajo con audio digital. Las obras fueron un intento por convertir a las mediaciones en un campo para la experimentación artística, buscando el uso no convencional y la apertura de éstas a la producción colaborativa con artistas y comunidades. Se concluyó que urge una mayor difusión escrita de las prácticas de experimentación sonora en México, que ha de promoverse una mayor reflexión sobre los mensajes imbuidos en una obra en función de sus mediaciones y que ha de desviarse a los medios para producir mensajes de libertad creativa.

La presencia de la radio en México se acerca a su centenario y a pesar de que la bibliografía publicada en torno a ella es abundante queda mucho por investigar sobre su pasado y su presente para intentar una prospectiva de su porvenir. En la investigación histórica nunca hay «última palabra»; los temas aparentemente ya estudiados siempre están abiertos a nuevas indagaciones generadoras de conocimiento.
Estas dos premisas se perciben en Homo Audiens III. Conocer la radio: Textos teóricos para aprehenderla, la nueva publicación del Seminario Permanente de Estudios sobre la Radio, coordinado por Virginia Medina Ávila, tercera entrega de una colección que ya ha contribuido de manera importante al estudio del fenómeno radiofónico.
Homo Audiens III nos entrega ocho textos con temas que van desde el uso gubernamental de la radio en la década de 1920 y llegan hasta el segundo decenio del siglo XXI momento en que la convergencia digital y la consolidación de un nuevo ecosistema mediático están definiendo los caminos por donde transitarán en los próximos años la radiodifusión sonora y el lenguaje creado por ella. En ese recorrido de prácticamente cien años, los autores de Homo Audiens III aportan datos novedosos y reflexiones sobre la radio y su radio y su relación con la política y con la tecnología, el papel de este medio como espacio para el entretenimiento cotidiano y como expresión artística, sus formas de consumo, las nuevas disposiciones que la Reforma en Telecomunicaciones y Radiodifusión de 2013 introdujo a la legislación mexicana aplicable a este medio de comunicación y los retos presentes y futuros de la radio en el entorno digital.

This paper is about a suite of electroacoustic music inspired by multiculturalism and DNA.  Samples, live playing, and synthetic sounds were combined using digital technology into a dance-informed, world-flavored, concert-oriented, acousmatic music.  Musique concrète, minimalism, rave, and hip hop all influenced the techniques, theories, and aesthetics used in composing the suite.  The decision to use the Ten Oxherding Pictures of Zen Buddhist fame as a non-diegetic enhancement of concert presentations impacted the poetics and provides a means by which to analyze the author’s own compositional process.  First the inspiration and potential issues are outlined, then the aesthetics are contextualized, the techniques used are described, and finally, the individual pieces of the suite are discussed.

We derive a novel explicit wave-domain model for “diode clipper” circuits with an arbitrary number of diodes in each orientation, applicable, e.g., to wave digital filter emulation of guitar distortion pedals. Improving upon and generalizing the model of Paiva et al. (2012), which approximates reverse-biased diodes as open circuits, we derive a model with an approximated correction term using two Lambert W functions. We study the energetic properties of each model and clarify aspects of the original derivation. We demonstrate the model’s validity by comparing a modded Tube Screamer clipping stage emulation to SPICE simulation.

The use of the laptop in performance causes various negative responses from the audience, who feel a loss of spectacle and performativity in the action. This occurs as a result of the lack of gesture and visual cues from the performer and the loss of focus from the audience, who have no visual object to ground their aural experience. A shift in focus from an understanding of the visual spectacle in performance to that of aural performativity is needed. Once this is understood, the audience can approach the performance with a shifted focus and come to new understandings of the aural object and contemporary digital audio.
KEYWORDS: laptop, aurality, contemporary digital audio, performativity, Pimmon

We present an analysis of the cowbell voice circuit from the Roland TR-808 Rhythm Composer. A digital model based on this analysis accurately emulates the original. Through the use of physical models of each sub-circuit, this model supports accurate emulation of circuit-bent extensions to the voice’s original behavior (including architecture-level alterations and component substitution). This behavior is very complicated and is inconvenient or impossible to capture accurately through black box modeling or structured sampling. The band pass filter sub-circuit is treated as a case study of applying Mason’s gain formula to finding the continuous-time transfer function of an analog circuit.

In the recent years, hybrid reverberation algorithms have been widely explored aiming to reproduce the acoustic behavior of real environment at low computational load. On this basis, exploiting the advantages introduced from hybrid reverberation structures, a novel approach for the reproduction of moving listener position through impulse responses (IR) interpolation has been presented in this paper. In particular, the presented methodology allows to remove redundant information in large IR database also decreasing the memory usage and the computational complexity required to perform the auralization operation. The effectiveness of the proposed approach has been proved taking into account a real IR database and also providing comparison with the existing state-of-art techniques in terms of objective and subjective measures.

In sound reproduction systems the audio crossover plays a fundamental role. Nowadays, digital crossover based on IIR filters are commonly employed, of which non-linear phase is a relevant topic. For this reason, solutions aiming to IIR filters approximating a linear phase behavior have been recently proposed. One of the latest exploits Fractional Derivative theory and uses Evolutionary Algorithms to explore the solution space in order to perform the IIR filter design: the IIR filter phase error is minimized to achieve a quasi-linear phase response. Nonetheless, this approach is not suitable for a crossover design, since the single filter transition band behavior is not predictable. This shoved the authors to propose a modified design technique including suitable constraints, as the amplitude response cutoff frequency, in the ad-hoc Particle Swarm Optimization algorithm exploring the space of IIR filter solutions. Simulations show that not only more performing filters can be obtained but also fully flat response crossovers achieved.

In the past years, several hybridization techniques have been proposed to synthesize novel audio content owing its properties from two audio sources. These algorithms, however, usually provide no feature learning, leaving the user, often intentionally, exploring parameters by trial-and-error. The introduction of machine learning algorithms in the music processing field calls for an investigation to seek for possible exploitation of their properties such as the ability to learn semantically meaningful features. In this first work, we adopt a Neural Network Autoencoder architecture and we enhance it to exploit temporal dependencies. In our experiments the architecture was able to modify the original timbre, resembling what it learned during the training phase, while preserving the pitch envelope from the input.

Aunque nació y fue etiquetado hace más de una década, el fenómeno del podcasting resurge hoy apoyado en la normalización del smartphone como dispositivo de consumo, en la versatilidad del audio para contar o reforzar historias y en la proliferación de agregadores y plataformas convencidas de las posibilidades para su monetización: es la era dorada del audio digital.

Artificial sound event detection (SED) has the aim to mimic the human ability to perceive and understand what is happening in the surroundings. Nowadays, deep learning offers valuable techniques for this goal such as convolutional neural networks (CNNs). The capsule neural network (CapsNet) architecture has been recently introduced in the image processing field with the intent to overcome some of the known limitations of CNNs, specifically regarding the scarce robustness to affine transformations (i.e., perspective, size, orientation) and the detection of overlapped images. This motivated the authors to employ CapsNets to deal with the polyphonic SED task, in which multiple sound events occur simultaneously. Specifically, we propose to exploit the capsule units to represent a set of distinctive properties for each individual sound event. Capsule units are connected through a so-called dynamic routing that encourages learning part-whole relationships and improves the detection performance in a polyphonic context. This paper reports extensive evaluations carried out on three publicly available datasets, showing how the CapsNet-based algorithm not only outperforms standard CNNs but also allows to achieve the best results with respect to the state-of-the-art algorithms.

The a.bel project aims to provide artists with a way to easily interact with their audience, making use of their participation to effectively craft unique performances. This paper gives an overview of the a.bel system and details the development of a suite of tools (as well as its integration into mobile applications) with which multimedia artists can easily create and distribute interactive content unto mobile devices.

Ante la normalización del smartphone como dispositivo dominante de acceso a la información en el entorno digital, la radio española ha asumido la necesidad de hacerse presente en estas pantallas con el fin de atraer y facilitar su escucha. Uno de los medios de mayor visibilidad y eficiencia lo constituyen las notificaciones push a través de las apps para móviles, cuyo diseño, contenido y estructura se analizan en este artículo. Se toman como muestra las alertas enviadas por las cuatro principales cadenas españolas de radio generalista de cobertura estatal –SER, COPE, Onda Cero y RNE– durante dos períodos no consecutivos de la temporada 2016/17 (enero-febrero y marzo-abril). El estudio evidencia el potencial de esta función para trasladar al móvil la inmediatez y proximidad de la radio e identifica los modelos periodísticos desarrollados por cada operador.

We introduce the technique of "Bit Bending," a particularly fertile technique for circuit bending which involves chort circuits and manipulations upon digital serial information. We present a justification for computer modeling of circuit-bent instruments, with deference to the movement's aversion to "theory-true" design and assocations with chance discovery. To facilitate software modeling of Bit Bending, we also present a software library for modeling certain classes of digital integrated circuits. A synthesis architecture case study (frequency modulation via numerically controlled oscillators) demonstrates software modeling of Bit Bending in action.

Based on the studies of Milner (2009), Katz (2004) and Wikstrom (2009), the article explores in details how the experience of listening and consuming music has been changing through the years due to the appearance and evolution of digital audio technology; analyzing the consequences in the recording industry and specially in its main product: popular music. The work approaches two different steps of this evolution (Compact Disks and Mp3) in terms of formats, devices and delivery methods; and how each one of them had an influence in our relationship with music.

Il phase vocoder (PV) è uno degli strumenti classici utilizzati nell’ambito digitale per l’analisi e la re-sintesi (synthesis by analysis) di uno spettro sonoro. L’obiettivo principale dell’analisi attraverso PV è quella di separare in maniera indipendente le informazioni temporali da quelle relative allo spettro di un segnale.
Il suo nome deriva storicamente dai primi vocoder (contrazione per “voice encoder”), i quali sviluppavano i concetti di Dudley e utilizzavano una batteria di bandpass filters attraverso i quali passava il segnale da analizzare. Il termine voice indica un primo utilizzo prettamente orientato ai segnali vocali e del parlato.
Nel 1966, James Flanagan e Roger Golden misero a punto una prima versione ai Bell Laboratories ma, rispetto alle intenzioni iniziali, il risultato non fu abbastanza soddisfacente. In seguito, negli anni ’70 venne sviluppata una versione più efficiente del PV implementando al suo interno la Short Time Fourier Transform.

Nous discuterons dans cet article de la définition de l'humain dans ses rapports actuels entretenus avec les environnements numériques, la nature et le traitement des données. Nos propos seront illustrés par les oeuvres des artistes numériques tels Ryoji Ikeda et son travail sur la visualisation des données ou celui du compositeur Zbigniew Karkowski sur la sonorisation des données. Une définition de l'homme--interfacé sera proposée. Elle montrera sa relation à un invisible devenu visible, à un inaudible devenu audible. Nous poursuivrons ensuite avec des exemples d'humanisation du non--humain. Une discussion théorique autour de l'homme--interfacé conclura notre propos.

Este libro es un intento de explicar qué es el audio digital desde varios puntos de vista: físico, lógico y musical y, además, con el doble propósito de hacerlo de manera divulgativa (que se entienda) y de manera práctica (que sirva para algo); sin pérdida de rigor.
Demasiados retos y ambiciones, pero aquí está. No es un manual, ni un libro técnico, ni un libro de teoría musical.
Es solo una introducción o aproximación al audio digital que quizá responda a alguna de sus dudas, justifique alguno de sus razonamientos, alimente su curiosidad o atraiga su atención hacia este maravilloso mundo donde las matemáticas se convierten en sonido, y viceversa.

The primary cause of injury-related death for the elders is represented by falls. The scientific community devoted them particular attention, since injuries can be limited by an early detection of the event. The solution proposed in this paper is based on a combined One-Class SVM (OCSVM) and template-matching classifier that discriminate human falls from nonfalls in a semisupervised framework. Acoustic signals are captured by means of a Floor Acoustic Sensor; then Mel-Frequency Cepstral Coefficients and Gaussian Mean Supervectors (GMSs) are extracted for the fall/nonfall discrimination. Here we propose a single-sensor two-stage user-aided approach: in the first stage, the OCSVM detects abnormal acoustic events. In the second, the template-matching classifier produces the final decision exploiting a set of template GMSs related to the events marked as false positives by the user. The performance of the algorithm has been evaluated on a corpus containing human falls and nonfall sounds. Compared to the OCSVM only approach, the proposed algorithm improves the performance by 10.14% in clean conditions and 4.84% in noisy conditions. Compared to Popescu and Mahnot (2009) the performance improvement is 19.96% in clean conditions and 8.08% in noisy conditions.

The watermarking of digital images, audio, video and multimedia products in general has been proposed for resolving copyright ownership and verifying originality of content. This paper studies the contribution of watermarking for developing protection schemes. A general watermarking framework (GWF) is studied and the fundamental demands are listed. The watermarking algorithms, namely watermark generation, embedding and detection, are analyzed and necessary conditions for a reliable and e cient protection are stated. Although the GWF satis es the majority of requirements for copyright protection and content veri cation, there are unsolved problems inside a pure watermarking framework. Particular solutions, based on product registration and related network services, are suggested to overcome such problems. The digital form of photographs, paintings, speech, music, video etc. became very popular in the last decade. Digital facilities for creating, processing and storing multimedia products have been found very convenient by creators, providers, editors and customers. At the same time, digital network communications have grown rapidly. In such an environment, digital products can be easily copied, processed for various purposes, broadcasted and/or publicly exposed. However, these revolutionary capabilities are also available to pirates who use them illegally for their personal interest by violating the legal rights of the providers and customers. Subsequently, security issues should be accounted for in the digital networked distribution systems for multimedia products. Digital piracy, dealing with multimedia products, generally, includes the following cases : Illegal access. A pirate tries to receive a digital product from a network site without permission. Intentional tampering. A pirate modi es a digital product in order to extract/insert features for malicious reasons and then proceeds to its retransmission. The authenticity of the original product is lost. Copyright violation. A pirate receives a product and resells it without getting the permission to do so from the copyright owner. Techniques based on cryptography, digital signatures and digital watermarks can be used for countering digital piracy 1]. Private or public key cryptography 2] can be used for data access control. Encrypted products are accessible, and decryption is possible only by someone who possesses a proper key. Well established algorithms (e.g. RSA 3] and DES 4]) can be used for this purpose. The encryption/decryption techniques should manipulate large amounts of digital data and should achieve real-time encryption/decryption e.g. for video and digital TV applications 1]. The

In this paper, we clarify what steganography is and what it can do. We contrast it with the related disciplines of cryptography and traffic security, present a unified terminology agreed at the first international workshop on the subject, and outline a number of approaches-many of them developed to hide encrypted copyright marks or serial numbers in digital audio or video. We then present a number of attacks, some new, on such information hiding schemes. This leads to a discussion of the formidable obstacles that lie in the way of a general theory of information hiding systems (in the sense that Shannon gave us a general theory of secrecy systems). However, theoretical considerations lead to ideas of practical value, such as the use of parity checks to amplify covertness and provide public key steganography. Finally, we show that public key information hiding systems exist, and are not necessarily constrained to the case where the warden is passive.

It has been programmed a propeller starter kit using SimpleIDE application to be a digital audio synthesizer. Generating signal of instruments is done by the keyboard as input and speaker as output to hear the sound. The method selection in processing of synthesis is performed within voltage controlled oscillator (VCO), voltage-controlled filter (VCF), analog amplifier, and amplitude envelope. The amplitudo envelopes used are attack, decay, sustain, and release (ADSR envelopes). This design known as substractive synthesis or an analogue synthesis.

Los podcasts narrativos de no ficción se han consolidado en 2021 como un género imprescindible en la oferta de productoras y plataformas de audio digital no sólo por el atractivo de sus argumentos e historias, sino porque renuevan las formas de contar el relato informativo y aportan credibilidad y prestigio a las redes que los distribuyen. En este artículo analizamos cómo el nuevo entorno de producción y consumo digital está favoreciendo la proliferación de un formato que recupera elementos expresivos menos empleados en la radio informativa al tiempo que exprime las ventajas creativas y la relación de intimidad con el oyente propias del audio bajo demanda.

Information hiding techniques have recently become important in a number of application areas. Digital audio, video, and pictures are increasingly furnished with distinguishing but imperceptible marks, which may contain a hidden copyright notice or serial number or even help to prevent unauthorised copying directly. Military communications systems make increasing use of traffic security techniques which, rather than merely concealing the content of a message using encryption, seek to conceal its sender, its receiver or its very existence. Similar techniques are used in some mobile phone systems and schemes proposed for digital elections. Criminals try to use whatever traffic security properties are provided intentionally or otherwise in the available communications systems, and police forces try to restrict their use. However, many of the techniques proposed in this young and rapidly evolving field can trace their history back to antiquity; and many of them are surprisingly easy to circumvent. In this article, we try to give an overview of the field; of what we know, what works, what does not, and what are the interesting topics for research.

Oversampling sigma-delta digital-to-analog converters are crucial building blocks for telecommunication applications. To reduce power consumption, lower oversampling ratios are preferred thus high-order digital sigma-delta modulators are needed to meet the dynamic performance requirements. This paper presents an oversampling DAC with 1.104 MHz signal bandwidth for DMT-ADSL application and focuses on the design issues of the high-order one-bit multiple feedback modulators (such as the stability problem, good inband SNDR performance, limit cycles, etc.). A new approach to obtain and optimize the stable feedback coefficients has been presented. From our analysis results it is found that the extra feedback coefficients and scaling coefficients in the modulator have non-negligible impact on the behavior of the limit cycles, and design guide for selecting the scaling coefficients is provided. Finally a 5th-order modulator with an oversampling ratio of 32 and 14-bit input has been implemented in a 0.6 μm 3.3 V CMOS process and integrated into the whole DAC chip.

A/D : Analog to Digital (da analogico a digitale) indica l'ingresso analogico di un convertitore audio digitale, per fare un esempio gli ingressi della vostra scheda audio o di un processore di effetti o di un mixer digitale, dopo lo stadio di preamplificazione hanno dei convertitori A/D. ADSR : acronimo di Attack, Decay Sustain e Release, è l'Envelope Generator (E.G.) il Generatore di Inviluppo presente anche in versioni ridotte su sintetizzatori, campionatori, batterie elettroniche ma anche su compressori, riverberi ecc. con le sue fasi di Attacco, Decadimento, Sostegno e Rilascio determina l'andamento nel tempo del parametro che sta controllando, decidendone quindi il tempo di risposta (Attack o Attack Time), la durata del picco dinamico (Decay), la durata del segnale (Sustain ma a volte Hold che sta per Tenuto) e la velocità di chiusura (Release). AIFF : acronimo di Audio Interchange File Format, uno dei formati dei file audio in uso soprattutto sui computer Apple. Aliasing : in un sistema di registrazione digitale, quando la frequenza da campionare supera quella consentita dalla frequenza di campionamento, contravvenendo al Teorema di Nyquist (che indica come la frequenza minima necessaria alla registrazione di un segnale sonoro sia pari al doppio della frequenza massima in esso contenuta) in cui i punti di quantizzazione non sono sufficienti a convertirla creando effetti indesiderati in zona udibile, vere e proprie distorsioni non presenti nel segnale analogico di origine, spesso se ne parla quando viene citato il filtro Anti-Aliasing , che serve appunto ad evitare questo grave inconveniente tramite l'applicazione al segnale di ingresso di una Filtro Passa Basso LPF (Low Pass Filter) onde evitare che giungano al convertitore A/D frequenze superiori alla metà della frequenza di campionamento. Bandwith : larghezza di banda; indica l'ampiezza di banda, in pratica le frequenze che possono passare attraverso un apparecchio o che vengono coinvolte da un filtro ad esempio di un Equalizzatore. Bass Reflex: è il sistema di caricamento del Woofer (l'altoparlante che si occupa specificamente delle basse frequenze) in una cassa acustica/monitor da studio. Consta in una o più aperture "accordate" della cassa, l'emissione è controllata nel tempo tramite l'impiego di materiale fonoassorbente e da una lunghezza ottimale del condotto d'aria, che rallentandone la propagazione riportano in fase le vibrazioni prodotte dalla superficie posteriore del Woofer,

A novel high capacity data hiding technique for digital audio is proposed. Imperceptibility of the embedded data is ensured based on the masking property of the human auditory system (HAS). Audio signal is decomposed into subband signals, some of which are selected for embedding data using finite-length impulse response approximations to allpass digital filters. Data detection is based on finding the filter pole-zero locations, which is achieved by power spectrum estimation of the data embedded audio signal. Performance of the proposed scheme is evaluated for different data encoding strategies. The proposed method is robust to desynchronization attacks as well as other standard data manipulation attacks.

A novel technique is proposed for data hiding in digital audio that exploits the low sensitivity of the human auditory system to phase distortion. Inaudible but controlled phase changes are introduced in the host audio using a set of allpass filters (APFs) with distinct parameters of allpass filters, i.e., pole-zero locations. The APF parameters are chosen to encode the embedding information. During the detection phase, the power spectrum of the audio data is estimated in the z-plane away from the unit circle. The power spectrum is used to estimate APF pole locations, for information decoding. Experimental results show that the proposed data hiding scheme can effectively withstand standard data manipulation attacks. Moreover, the proposed scheme is shown to embed 5-8 times more data than the existing audio data hiding schemes while providing comparable perceptual performance and robustness

A novel perception-based data hiding technique for digital audio is proposed. It exploits lower sensitivity of human auditory system (HAS) to phase distortion in audio compared with magnitude distortion. Audio is decomposed into subband signals some of which are selected for embedding data with a controlled alteration of phase using suitable allpass digital filters. The proposed scheme is robust to standard data manipulations yielding less than 2% error probability against compression, re-sampling, re-quantization, random chopping and noise addition. The proposed method is also robust to the desynchronization attacks.